1. Technical Field
The present invention relates to a noise reduction system and the method thereof for a passenger compartment of automotive vehicle by positively generating a sound from a sound source to cancel the vehicle internal noise and more specifically relates to a noise reduction system to reduce a noise produced periodically.
2. Related Prior Arts
There have been proposed several techniques for reducing a noise sound in the passenger compartment by producing a canceling sound having the same amplitude as the noise sound and a reversed phase thereto from a sound source disposed in the passenger compartment.
Among these prior arts, Japanese Patent Application Laid Open No. Toku-Kai-Hei 3-178846 discloses the following technique:
Referring now to FIGS. 9 to 11, FIG. 9 is a block diagram of the noise reduction system according to the prior art. FIG. 10 is a block diagram showing the adaptive filter section and the tap value updating section according to the prior art. Further, FIG. 11 is a block diagram showing the the prior art. In FIG. 9, numeral 10 indicates a noise source, numeral 11 a pick-up circuit, numerals 12 and 16 analogue-to-digital (A/D) converters, numeral digital-to-analogue (D/A) converter, numeral 14 a speaker, numeral 7 an adaptive filter, numeral 8 a transmission characteristic compensation section and numeral 9 a tap value updating section.
A microphone 15 is disposed in a position where a noise is to be reduced. The adaptive filter 7 corrects an error signal e (t), namely a difference between a noise signal picked up by the pick-up circuit 11 and a noise inputted into the microphone 15 and the corrected signal is transmitted from the speaker 14. Then the signal which reaches the microphone 15 generates a signal having the same amplitude as and a reversed phase to the noise sound from the noise source 10.
As details will be described hereinafter in FIG. 10, the adaptive filter 7 is a digital filter composed of delay lines with tap. Namely, by inputting an output signal from the pick-up circuit to the adaptive filter 7, the transmission characteristic of filter can be determined such that a sound pressure and a wave form are reversed at the position of the microphone 15. This adaptation is performed at the tap value updating section 9.
Since compensating transmission characteristics are affected by a time lag, a band restriction or the like while a signal is generated from the adaptive filter 7 and reaches the microphone 15 through the D/A converter 13 and the speaker 14, the transmission characteristic compensation section 8 act as compensating them and sending a compensated signal having the same amplitude as and the reversed phase to the signal from the noise source 10 to the tap value updating section 9.
These transmission characteristics can also be composed of digital filters of delay lines with tap. FIG. 11 is a schematic diagram showing a composition of the transmission characteristic compensation section 8. Numerals 80-1 to 80-J are delay elements for controlling a timing of sampling pulse inputted to the A/D converters 12 and 16. Further, 81-0 to 81-J are tap values by which an output value of the delay element is multiplied and then the multiplied output value is outputted.
Now where the output value of the A/D converter 12 is x (n) at t=t.sub.n and x (n+1) at t=t.sub.n+1, further where &lt;i=1, 3&gt;.SIGMA.x.sub.i =x.sub.1 +x.sub.2 +x.sub.3 is expressed, the compensation signal C (n) from the transmission characteristic compensation section 8 is expressed as EQU C(n)=&lt;i=0, J&gt;.SIGMA.x (n-i)C.sub.i ( 1)
The adaptive filter 7 comprises delay elements 70-1 to 70-Z, tap values 71-0 to 71-Z and an adder 72, as shown in FIG. 10, The delay element 70 controls a timing of sampling pulse inputted to the A/D converter 12.
Consequently, the output y (n) from the adaptive filter is expressed as: EQU y(n)=&lt;i=0,Z&gt;.SIGMA.x(n-i)W.sub.i (n) (2)
y (n) is converted into an analogue signal in the D/A converter 13 and transmitted to the speaker 14.
The tap values W.sub.O (n) to W.sub.z (n) of the adaptive filter 7 are updated at the tap value updating section 9 each time the sampling pulse is generated. As illustrated in FIG. 10, the tap value updating section 9 comprises multipliers 90, 91 and 92 and an adder 93.
First, in the delay element 90 the output signal C(n) from the transmission characteristic compensation section 8 inputted and transmitted after the signal is delayed by a time equal to the sampling pulse interval. Further, in the multiplier the output e(t) from the microphone 15 is multiplied by .alpha. after being converted into a digital value in the A/D converter 16. This value .alpha. is predetermined according to a loop characteristic of the adaptive control system.
Next, the updated W (n+1) is calculated with respect to each of tap values of the adaptive filter 7. The explanation will be made about the case where the tap value W.sub.0 (n) of the tap 71-0 is updated to W.sub.O (n+1) in order to make explanation easier. In the multiplier 92-0 the output of the multiplier 91 is multiplied by the output value C (n) of the transmission characteristic compensation section 8. the adder 93-0 the output value of the multiplier 92-0 is subtracted from the tap value W.sub.0 (n) at t=t.sub.n and the result of the subtraction is updated into a tap value W.sub.0 (n+1) at the next t=t.sub.n+1, That is to say: EQU W.sub.0 (n+1)=W.sub.0 (n)-.alpha.C(n)e(n) (3)
Further, other tap value Wi will be updated as follows; EQU W.sub.i (n+1)=W.sub.i (n)-.alpha.C(n-i)e(n) (4)
As described hereinbefore, in the noise reduction system according to the prior art by means of passing a noise signal picked up from the noise source through the adaptive filter a sound having the same amplitude as and a reversed phase to a noise sound is generated from the speaker 14 to reduce the noise in the vicinity of the microphone.
Accordingly, it is necessary to carry out equal number of multiplications in the adaptive filter and equal number of additions to tap numbers in the tap value updating section.
When these multiplications and additions are carried out by independent multipliers and adders, a construction of the system becomes very complicated, therefore commonly these calculations are performed by a processor. However, even when using a processor, in order to carry out equal number of multiplications and additions to tap numbers at an interval of sampling pulse an expensive high speed processor is needed.